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Médias (1)
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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
Autres articles (17)
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Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
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Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)
Sur d’autres sites (3426)
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Qt Video Recorder
11 mai 2014, par DavlogI am trying to create a video recorder with Qt. What I did so far was taking a screenshot of a rectangle on the screen and save it. At the end I use ffmpeg to get a video file out of the images.
I connected a timer’s signal
timeout()
to my custom slot which takes the snapshot and saves it to my tmp folder. The timer has an intervall of 1000 / 30. That should be 30 times per second. But 1000 / 30 is a little bit more than 33 milliseconds so I cannot really get 30 fps. It’s a bit more.I recorded a youtube video with my recorder and everything was smooth but a little bit faster / slower depending on the intervall.
So my question basically is how do I get really 30 / 40 / 50 / ... fps ?
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how to convert an MPEGTS file into an FLV file by programming with libavcodec
9 mai 2014, par Hexing BI wanna convert an MPEGTS file into an FLV file with libavcodec APIs(just transformat, video/audio codecs are not changed). Following is the code I found on web. After some hack, it can generate an FLV file, which could not be played. Any clue to fix this ?
#include
#include
#include
#include
#include <libavformat></libavformat>avformat.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libavutil></libavutil>avutil.h>
#include <libavutil></libavutil>rational.h>
#include <libavdevice></libavdevice>avdevice.h>
#include <libavutil></libavutil>mathematics.h>
#include <libswscale></libswscale>swscale.h>
static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream) {
AVCodecContext* input_codec_context = NULL;
AVCodecContext* output_codec_context = NULL;
AVStream* output_stream = NULL;
output_stream = avformat_new_stream(output_format_context, 0);
if (!output_stream) {
printf("Call av_new_stream function failed\n");
return NULL;
}
input_codec_context = input_stream->codec;
output_codec_context = output_stream->codec;
output_codec_context->codec_id = input_codec_context->codec_id;
output_codec_context->codec_type = input_codec_context->codec_type;
// output_codec_context->codec_tag = input_codec_context->codec_tag;
output_codec_context->codec_tag = av_codec_get_tag(output_format_context->oformat->codec_tag, input_codec_context->codec_id);
output_codec_context->bit_rate = input_codec_context->bit_rate;
output_codec_context->extradata = input_codec_context->extradata;
output_codec_context->extradata_size = input_codec_context->extradata_size;
if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000) {
output_codec_context->time_base = input_codec_context->time_base;
output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
} else {
output_codec_context->time_base = input_stream->time_base;
}
switch (input_codec_context->codec_type) {
case AVMEDIA_TYPE_AUDIO:
output_codec_context->channel_layout = input_codec_context->channel_layout;
output_codec_context->sample_rate = input_codec_context->sample_rate;
output_codec_context->channels = input_codec_context->channels;
output_codec_context->frame_size = input_codec_context->frame_size;
if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3) {
output_codec_context->block_align = 0;
} else {
output_codec_context->block_align = input_codec_context->block_align;
}
break;
case AVMEDIA_TYPE_VIDEO:
output_codec_context->pix_fmt = input_codec_context->pix_fmt;
output_codec_context->width = input_codec_context->width;
output_codec_context->height = input_codec_context->height;
output_codec_context->has_b_frames = input_codec_context->has_b_frames;
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER) {
output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
// output_codec_context->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
break;
default:
break;
}
return output_stream;
}
int main(int argc, char* argv[]) {
const char* input = argv[1];
const char* output_prefix = NULL;
char* segment_duration_check = 0;
const char* index = NULL;
char* tmp_index = NULL;
const char* http_prefix = NULL;
long max_tsfiles = 0;
double prev_segment_time = 0;
double segment_duration = 0;
AVInputFormat* ifmt = NULL;
AVOutputFormat* ofmt = NULL;
AVFormatContext* ic = NULL;
AVFormatContext* oc = NULL;
AVStream* video_st = NULL;
AVStream* audio_st = NULL;
AVCodec* codec = NULL;
AVDictionary* pAVDictionary = NULL;
av_register_all();
av_log_set_level(AV_LOG_DEBUG);
char szError[256] = {0};
int nRet = avformat_open_input(&ic, input, ifmt, &pAVDictionary);
if (nRet != 0) {
av_strerror(nRet, szError, 256);
printf(szError);
printf("\n");
printf("Call avformat_open_input function failed!\n");
return 0;
}
if (avformat_find_stream_info(ic, NULL) < 0) {
printf("Call av_find_stream_info function failed!\n");
return 0;
}
ofmt = av_guess_format(NULL, argv[2], NULL);
if (!ofmt) {
printf("Call av_guess_format function failed!\n");
return 0;
}
oc = avformat_alloc_context();
if (!oc) {
printf("Call av_guess_format function failed!\n");
return 0;
}
oc->oformat = ofmt;
int video_index = -1, audio_index = -1;
unsigned int i;
for (i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++) {
switch (ic->streams[i]->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
video_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
video_st = add_output_stream(oc, ic->streams[i]);
if (video_st->codec->codec_id == CODEC_ID_H264) {
video_st->codec->opaque = av_bitstream_filter_init("h264_mp4toannexb");
}
break;
case AVMEDIA_TYPE_AUDIO:
audio_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
audio_st = add_output_stream(oc, ic->streams[i]);
if (audio_st->codec->codec_id == CODEC_ID_AAC && !audio_st->codec->extradata_size) {
audio_st->codec->opaque = av_bitstream_filter_init("aac_adtstoasc");
}
break;
default:
ic->streams[i]->discard = AVDISCARD_ALL;
break;
}
}
codec = avcodec_find_decoder(video_st->codec->codec_id);
if (codec == NULL) {
printf("Call avcodec_find_decoder function failed!\n");
return 0;
}
if (avcodec_open2(video_st->codec, codec, NULL) < 0) {
printf("Call avcodec_open function failed !\n");
return 0;
}
if (avio_open(&oc->pb, argv[2], AVIO_FLAG_WRITE) < 0) {
return 0;
}
if (avformat_write_header(oc, &pAVDictionary)) {
printf("Call avformat_write_header function failed.\n");
return 0;
}
int decode_done = 0;
do {
AVPacket packet;
decode_done = av_read_frame(ic, &packet);
if (decode_done < 0) {
break;
}
if (packet.stream_index == video_index && (packet.flags & AV_PKT_FLAG_KEY) && video_st->codec->opaque != NULL) {
AVPacket pkt = packet;
int a = av_bitstream_filter_filter(video_st->codec->opaque, video_st->codec, NULL, &pkt.data, &pkt.size,
packet.data, packet.size, packet.flags & AV_PKT_FLAG_KEY);
if (a == 0) {
memmove(packet.data, pkt.data, pkt.size);
packet.size = pkt.size;
} else if (a > 0) {
packet = pkt;
}
}
else if (packet.stream_index == audio_index && audio_st->codec->opaque != NULL) {
AVPacket pkt = packet;
int a = av_bitstream_filter_filter(audio_st->codec->opaque, audio_st->codec, NULL, &pkt.data, &pkt.size,
packet.data, packet.size, packet.flags & AV_PKT_FLAG_KEY);
if (a == 0) {
memmove(packet.data, pkt.data, pkt.size);
packet.size = pkt.size;
} else if (a > 0) {
packet = pkt;
}
}
nRet = av_interleaved_write_frame(oc, &packet);
if (nRet < 0) {
printf("Call av_interleaved_write_frame function failed\n");
} else if (nRet > 0) {
printf("End of stream requested\n");
av_free_packet(&packet);
break;
}
av_free_packet(&packet);
} while(!decode_done);
av_write_trailer(oc);
av_bitstream_filter_close(video_st->codec->opaque);
av_bitstream_filter_close(audio_st->codec->opaque);
avcodec_close(video_st->codec);
unsigned int k;
for(k = 0; k < oc->nb_streams; k++) {
av_freep(&oc->streams[k]->codec);
av_freep(&oc->streams[k]);
}
av_free(oc);
getchar();
return 0;
} -
How to merge audio and video files with specific frame rate with ffmpeg ? [on hold]
3 mai 2014, par MulagalaI want to merge to files, video file is ’one.webm’,and audio file is ’def.wav’.I have successfully merged these to files to a avi format.But i am getting 1000 frames per second.How to reduce the frame rate.I used this command to merge
ffmpeg -i def.wav -i abc.webm -acodec copy -vcodec copy output.avi
i am using
linux fedora19
Thanks in advance !